Traditionally, telephone calls have been routed exclusively through the public switched telephone network (PSTN) which is based on time division multiplexing (TDM) circuit switches with TDM trunk groups interconnecting the circuit switches. In recent years, however, Internet telephony has become increasingly popular. Voice calls, whether dialed on a landline telephone or mobile device or placed from a computer, can now be routed at least in part over the Internet, which is primarily a packet-switching network, based on one or more voice over Internet Protocol (VoIP) protocols.
Session Initiation Protocol (SIP) is a prevalent VoIP protocol developed by the Internet Engineering Task Force (IETF). SIP is an application-layer signaling protocol for creating, modifying, and terminating peer-to-peer sessions with one or more participants. These sessions are not limited to voice calls but may also include multimedia distribution and multimedia conferences.
Since the PSTN still exists and is expected to continue carrying a significant portion of voice calls, it is necessary to provide seamless interoperability between SIP and the PSTN, such that voice calls can be routed fluently between the PSTN and the Internet. For example, in the PSTN, voice calls are typically routed and processed according to their respective TDM trunk groups, which provide contextual information (or service profile) for each call. When a voice call is routed between the PSTN and the Internet, it may be desirable to preserve relevant context information associated with that voice call.
FIG. 1 illustrates current co-existence of the PSTN and the Internet and alternative call routing methods. The PSTN may be divided into different areas (e.g., PSTN Area 1 and PSTN Area 2) each with multiple callers connected thereto. Traditionally, a telephone call from Caller 1 to Caller 5 may be routed from PSTN Area 1 to PSTN Area 2 via a TDM trunk group. Nowadays, the call from Caller 1 to Caller 5 may more likely be routed through one or more SIP switches 102 (or switches based on other VoIP protocols such as H.323) over the Internet, and then back to the PSTN (e.g., PSTN Area 2), before the call eventually reaches Caller 5. The detour via the alternative route requires that the relevant VoIP protocol (e.g., SIP) to seamlessly interoperate with the PSTN in order to properly process and/or route the telephone call.
However, existing approaches for SIP-PSTN interoperability have achieved only limited success. According to one approach, a proprietary SIP header can be included in SIP signaling messages to communicate trunk group information between a pair of SIP nodes. Unfortunately, such a proprietary header is unlikely to see widespread use or acceptance. Another approach has been discussed in an IETF Internet draft entitled “Representing trunk groups in tel/sip Uniform Resource Identifiers (URIs).” According to this approach, trunk group parameters may be structured and represented as a standard extension to the tel URI. There are a few shortcomings with this approach. For example, this approach focuses only on edge switches that are connected with ingress or egress TDM trunk groups, and it does not address communications between a pair of internal SIP nodes. The mechanism of using tel URIs to carry TDM trunk group parameters does not facilitate any traffic segregation between two internal SIP nodes. For traffic segregation, it has been proposed that multiple pairs of Internet Protocol (IP) addresses can be assigned to two SIP nodes, one pair for each sub-channel between the two SIP nodes. This approach will significantly increase the number of IP addresses required for each SIP node, which is impracticable and undesirable.
In view of the foregoing, it would be desirable to provide a solution for SIP-PSTN interoperability which overcomes the above-described inadequacies and shortcomings.